There is no such thing as a "SIP Trunk" in Asterisk. Nope. It does not exist. Some people (seems to me mostly GUI people) use the term "SIP trunk" to mean "SIP friend/user/peer".
John covici wrote: > I am not an expert on chanspy, but it seems to me spying on the trunk > would not work very well, would not you hear multiple conversations > mixed if more than one extension were calling? Seems best to me to > spy on an extension. YOu also can do a show channels to see who is > talking to whom. > > on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote > > The parameter to Chanspy should be the whole or part of the channel name. > I do not understand what you mean by "sip trunk". It make perfect sense that > you can hear both streams of voice when you use the phone's extension as > Asterisk usually uses "SIP/extension+xxx" as the channel name of the call. > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] on behalf of Ed Nuñez > > Sent: Wed 9/26/2007 4:48 PM > > To: [EMAIL PROTECTED] > > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: Re: [asterisk-users] ChanSpy issue > > > > > > > > Hello list > > > > > > > > I am having an issue with Chanspy/SIP that I'm hoping someone has come > > across and resolved in the past. > > > > > > > > I am sending calls that come in TDM through T1 ZAP channels and go out to a > > SIP trunk. > > > > > > > > If I spy on the SIP channel, I can hear the person on the SIP side of the > > call just fine, but the person on the ZAP channel fades in and out. > > > > If I spy on the ZAP channel, and can hear both sides just fine, but I don't > > know who I am spying on since I have other calls coming in on the same T1. > > > > > > > > If I spy on a SIP extension instead of a SIP trunk, I hear both sides just > > fine. > > > > > > > > I am using a recent version of Asterisk 1.2 and I am using g729 licenses. > > > > > > > > This is the command I am using to spy. > > > > > > > > exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) > > > > > > > > > > > > > > > > > > > > > > <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN"> > > <HTML> > > <HEAD> > > <META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"> > > <META NAME="Generator" CONTENT="MS Exchange Server version 6.5.7638.1"> > > <TITLE>RE: [asterisk-users] ChanSpy issue</TITLE> > > </HEAD> > > <BODY> > > <!-- Converted from text/plain format --> > > > > <P><FONT SIZE=2>The parameter to Chanspy should be the whole or part of > the channel name. I do not understand what you mean by "sip trunk". > It make perfect sense that you can hear both streams of voice when you use > the phone's extension as Asterisk usually uses "SIP/extension+xxx" > as the channel name of the call.<BR> > > <BR> > > <BR> > > -----Original Message-----<BR> > > From: [EMAIL PROTECTED] on behalf of Ed Nuñez<BR> > > Sent: Wed 9/26/2007 4:48 PM<BR> > > To: [EMAIL PROTECTED]<BR> > > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'<BR> > > Subject: Re: [asterisk-users] ChanSpy issue<BR> > > <BR> > > <BR> > > <BR> > > Hello list<BR> > > <BR> > > <BR> > > <BR> > > I am having an issue with Chanspy/SIP that I'm hoping someone has come<BR> > > across and resolved in the past.<BR> > > <BR> > > <BR> > > <BR> > > I am sending calls that come in TDM through T1 ZAP channels and go out to > a<BR> > > SIP trunk.<BR> > > <BR> > > <BR> > > <BR> > > If I spy on the SIP channel, I can hear the person on the SIP side of > the<BR> > > call just fine, but the person on the ZAP channel fades in and out.<BR> > > <BR> > > If I spy on the ZAP channel, and can hear both sides just fine, but I > don't<BR> > > know who I am spying on since I have other calls coming in on the same > T1.<BR> > > <BR> > > <BR> > > <BR> > > If I spy on a SIP extension instead of a SIP trunk, I hear both sides > just<BR> > > fine.<BR> > > <BR> > > <BR> > > <BR> > > I am using a recent version of Asterisk 1.2 and I am using g729 > licenses.<BR> > > <BR> > > <BR> > > <BR> > > This is the command I am using to spy.<BR> > > <BR> > > <BR> > > <BR> > > exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))<BR> > > <BR> > > <BR> > > <BR> > > <BR> > > <BR> > > <BR> > > <BR> > > <BR> > > <BR> > > <BR> > > </FONT> > > </P> > > > > </BODY> > > </HTML>_______________________________________________ > > > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
